With steady growth of access bandwidth, Internet applications are increasingly starting to use streaming audio and video contents. Since the current Internet is inherently a heterogeneous and dynamic best-effort network, channel bandwidth usually fluctuates in a wide range from bit rate below 64 kbps to well above 1 Mbps. This brings great challenges to video coding and streaming technologies in providing a smooth playback experience and best available video quality to the users.
By way of example, conventional streamed video, for example, over the Internet, is transmitted with TCP or UDP with TFRC. One problem with this is that there can be a sharp reduction in the sending rate, e.g., by half, when packet loss occurs. This is typically an outcome of the rate control mechanism, which prevents the Internet from fatal congestion collapse. However, such a sharp sending rate reduction can lead to significant quality reduction, which is quite undesirable for streaming video.
In addition to various and often frequent bandwidth fluctuations, current Internet streaming applications tend to lack accurate estimations of the available network/channel bandwidth. As such, the applications are often unable to make full use of the available network resources and/or actively adjust the sending rate when the network gets congested.
Hence, there is a need for improved methods and apparatuses for use in the Internet and other like networks and with streaming media applications.